[Asterisk-video] Asterisk H264

Chaim Fried cef at google.com
Wed Jul 5 14:38:14 MST 2006 

On Jul 5, 2006, at 5:24 PM, Chaim Fried wrote:

> i am on the latest trunk....
> i am using tandberg video endpoints. they work fine with Asterisk  
> on h.263 but not h.264..
> here is a sip debug snippet...
>
> nyguest25*CLI> sip debug
> SIP Debugging enabled
> Retransmitting #6 (no NAT) to zzz.zzz.zzz.zzz:5060:
> NOTIFY sip:zzz.zzz.zzz.zzz SIP/2.0
> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK4313f6e2;rport
> From: "asterisk" <sip:asterisk at aaa.aaa.aaa.aaa>;tag=as1db072c9
> To: <sip:zzz.zzz.zzz.zzz>
> Contact: <sip:asterisk at aaa.aaa.aaa.aaa>
> Call-ID: 0f8522eb2ae0624b13e179fb582023de at aaa.aaa.aaa.aaa
> CSeq: 102 NOTIFY
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Event: message-summary
> Content-Type: application/simple-message-summary
> Content-Length: 93
>
> Messages-Waiting: no
> Message-Account: sip:asterisk at aaa.aaa.aaa.aaa
> Voice-Message: 0/0 (0/0)
>
> ---
> Retransmitting #6 (no NAT) to zzz.zzz.zzz.zzz:5060:
> NOTIFY sip:zzz.zzz.zzz.zzz SIP/2.0
> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK4313f6e2;rport
> From: "asterisk" <sip:asterisk at aaa.aaa.aaa.aaa>;tag=as1db072c9
> To: <sip:zzz.zzz.zzz.zzz>
> Contact: <sip:asterisk at aaa.aaa.aaa.aaa>
> Call-ID: 0f8522eb2ae0624b13e179fb582023de at aaa.aaa.aaa.aaa
> CSeq: 102 NOTIFY
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Event: message-summary
> Content-Type: application/simple-message-summary
> Content-Length: 93
>
> Messages-Waiting: no
> Message-Account: sip:asterisk at aaa.aaa.aaa.aaa
> Voice-Message: 0/0 (0/0)
>
> ---
> nyguest25*CLI>
> <-- SIP read from xxx.xxx.xxx.xxx:5060:
> INVITE sip:3001 at aaa.aaa.aaa.aaa SIP/2.0
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bKa4c54c13aa2c5f75f3b1a611eaa51cd8.1;rport
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 100 INVITE
> Contact: <sip:3005 at xxx.xxx.xxx.xxx>
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>
> Max-Forwards: 70
> Allow: UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
> User-Agent: TANDBERG/46 (F3.3 NTSC)
> Content-Type:application/sdp
> Content-Length:433
>
> v=0
> o=tandberg 0 1 IN IP4 xxx.xxx.xxx.xxx
> s=-
> c=IN IP4 xxx.xxx.xxx.xxx
> b=CT:512
> t=0 0
> m=audio 5600 RTP/AVP 96 97 9 8 0
> c=IN IP4 xxx.xxx.xxx.xxx
> a=sendrecv
> a=rtpmap:96 G7221/16000
> a=fmtp:96 bitrate=32000
> a=rtpmap:97 G7221/16000
> a=fmtp:97 bitrate=24000
> a=rtpmap:9 G722/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=video 5602 RTP/AVP 31
> c=IN IP4 xxx.xxx.xxx.xxx
> a=sendrecv
> a=rtpmap:31 H261/90000
> a=fmtp:31 cif=1;qcif=1
>
> --- (12 headers 21 lines)---
>
> <-- SIP read from xxx.xxx.xxx.xxx:5060:
> INVITE sip:3001 at aaa.aaa.aaa.aaa SIP/2.0
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bKa4c54c13aa2c5f75f3b1a611eaa51cd8.1;rport
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 100 INVITE
> Contact: <sip:3005 at xxx.xxx.xxx.xxx>
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>
> Max-Forwards: 70
> Allow: UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
> User-Agent: TANDBERG/46 (F3.3 NTSC)
> Content-Type:application/sdp
> Content-Length:433
>
> v=0
> o=tandberg 0 1 IN IP4 xxx.xxx.xxx.xxx
> s=-
> c=IN IP4 xxx.xxx.xxx.xxx
> b=CT:512
> t=0 0
> m=audio 5600 RTP/AVP 96 97 9 8 0
> c=IN IP4 xxx.xxx.xxx.xxx
> a=sendrecv
> a=rtpmap:96 G7221/16000
> a=fmtp:96 bitrate=32000
> a=rtpmap:97 G7221/16000
> a=fmtp:97 bitrate=24000
> a=rtpmap:9 G722/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=video 5602 RTP/AVP 31
> c=IN IP4 xxx.xxx.xxx.xxx
> a=sendrecv
> a=rtpmap:31 H261/90000
> a=fmtp:31 cif=1;qcif=1
>
> --- (12 headers 21 lines)---
> Sending to xxx.xxx.xxx.xxx : 5060 (NAT)
> Using INVITE request as basis request -  
> 9c00870094007300 at xxx.xxx.xxx.xxx
> Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bKa4c54c13aa2c5f75f3b1a611eaa51cd8.1;rport;received=x 
> xx.xxx.xxx.xxx
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>;tag=as220a6b2b
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 100 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:3001 at aaa.aaa.aaa.aaa>
> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",  
> nonce="33eeb7ce"
> Content-Length: 0
>
>
> ---
> Sending to xxx.xxx.xxx.xxx : 5060 (NAT)
> Using INVITE request as basis request -  
> 9c00870094007300 at xxx.xxx.xxx.xxx
> Scheduling destruction of SIP dialog  
> '9c00870094007300 at xxx.xxx.xxx.xxx' in 32000 ms (Method: INVITE)
> Found user '3005'
> Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bKa4c54c13aa2c5f75f3b1a611eaa51cd8.1;rport;received=x 
> xx.xxx.xxx.xxx
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>;tag=as220a6b2b
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 100 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> ontact: <sip:3001 at aaa.aaa.aaa.aaa>
> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",  
> nonce="33eeb7ce"
> Content-Length: 0
>
>
> ---
> Scheduling destruction of SIP dialog  
> '9c00870094007300 at xxx.xxx.xxx.xxx' in 32000 ms (Method: INVITE)
> Found user '3005'
> nyguest25*CLI>
> <-- SIP read from xxx.xxx.xxx.xxx:5060:
> ACK sip:3001 at aaa.aaa.aaa.aaa SIP/2.0
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bKa4c54c13aa2c5f75f3b1a611eaa51cd8.1;rport
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 100 ACK
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>;tag=as220a6b2b
> User-Agent: TANDBERG/46 (F3.3 NTSC)
> Content-Length:0
>
>
> --- (8 headers 0 lines)---
>
> <-- SIP read from xxx.xxx.xxx.xxx:5060:
> ACK sip:3001 at aaa.aaa.aaa.aaa SIP/2.0
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bKa4c54c13aa2c5f75f3b1a611eaa51cd8.1;rport
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 100 ACK
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>;tag=as220a6b2b
> User-Agent: TANDBERG/46 (F3.3 NTSC)
> Content-Length:0
>
>
> --- (8 headers 0 lines)---
> nyguest25*CLI>
> <-- SIP read from xxx.xxx.xxx.xxx:5060:
> INVITE sip:3001 at aaa.aaa.aaa.aaa SIP/2.0
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bK184b70b8ed4c2761877486319683c457.1;rport
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 101 INVITE
> Contact: <sip:3005 at xxx.xxx.xxx.xxx>
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>
> Max-Forwards: 70
> Allow: UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
> User-Agent: TANDBERG/46 (F3.3 NTSC)
> Proxy-Authorization: Digest nonce="33eeb7ce", realm="asterisk",  
> qop="", username="3005", uri="sip:aaa.aaa.aaa.aaa",  
> response="feea40f87fa828da3280acf2a7381e01", algorithm=MD5
> Content-Type:application/sdp
> Content-Length:433
>
> v=0
> o=tandberg 0 1 IN IP4 xxx.xxx.xxx.xxx
> s=-
> c=IN IP4 xxx.xxx.xxx.xxx
> b=CT:512
> t=0 0
> m=audio 5600 RTP/AVP 96 97 9 8 0
> c=IN IP4 xxx.xxx.xxx.xxx
> a=sendrecv
> a=rtpmap:96 G7221/16000
> a=fmtp:96 bitrate=32000
> a=rtpmap:97 G7221/16000
> a=fmtp:97 bitrate=24000
> a=rtpmap:9 G722/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=video 5602 RTP/AVP 31
> c=IN IP4 xxx.xxx.xxx.xxx
> a=sendrecv
> a=rtpmap:31 H261/90000
> a=fmtp:31 cif=1;qcif=1
>
> --- (13 headers 21 lines)---
> Sending to xxx.xxx.xxx.xxx : 5060 (NAT)
> Using INVITE request as basis request -  
> 9c00870094007300 at xxx.xxx.xxx.xxx
> Found user '3005'
> Found RTP audio format 96
> Found RTP audio format 97
> Found RTP audio format 9
> Found RTP audio format 8
> Found RTP audio format 0
> Found RTP video format 31
> Peer audio RTP is at port xxx.xxx.xxx.xxx:5600
> Found description format G7221 for ID 96
> Found description format G7221 for ID 97
> Found description format G722 for ID 9
> Found description format PCMA for ID 8
> Found description format PCMU for ID 0
> Found description format H261 for ID 31
> Capabilities: us - 0x28070f (g723|gsm|ulaw|alaw|g729|speex|ilbc| 
> h263|h264), peer - audio=0x4040c (ulaw|alaw|ilbc|h261)/ 
> video=0x40000 (h261), combined - 0x40c (ulaw|alaw|ilbc)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer -  
> 0x0 (nothing), combined - 0x0 (nothing)
> Peer audio RTP is at port xxx.xxx.xxx.xxx:5600
> Peer video RTP is at port xxx.xxx.xxx.xxx:5602
> Looking for 3001 in from-sip (domain aaa.aaa.aaa.aaa)
> list_route: hop: <sip:3005 at xxx.xxx.xxx.xxx>
>
> <-- SIP read from xxx.xxx.xxx.xxx:5060:
> INVITE sip:3001 at aaa.aaa.aaa.aaa SIP/2.0
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bK184b70b8ed4c2761877486319683c457.1;rport
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 101 INVITE
> Contact: <sip:3005 at xxx.xxx.xxx.xxx>
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>
> Max-Forwards: 70
> Allow: UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
> User-Agent: TANDBERG/46 (F3.3 NTSC)
> Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bK184b70b8ed4c2761877486319683c457.1;rport;received=x 
> xx.xxx.xxx.xxx
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:3001 at aaa.aaa.aaa.aaa>
> Content-Length: 0
>
>
> ---
> e="33eeb7ce", realm="asterisk", qop="", username="3005",  
> uri="sip:aaa.aaa.aaa.aaa",  
> response="feea40f87fa828da3280acf2a7381e01", algorithm=MD5
> Content-Type:application/sdp
> Content-Length:433
>
> v=0
> o=tandberg 0 1 IN IP4 xxx.xxx.xxx.xxx
> s=-
> c=IN IP4 xxx.xxx.xxx.xxx
> b=CT:512
> t=0 0
> m=audio 5600 RTP/AVP 96 97 9 8 0
> c=IN IP4 xxx.xxx.xxx.xxx
> a=sendrecv
> a=rtpmap:96 G7221/16000
> a=fmtp:96 bitrate=32000
> a=rtpmap:97 G7221/16000
> a=fmtp:97 bitrate=24000
> a=rtpmap:9 G722/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> m=video 5602 RTP/AVP 31
> c=IN IP4 xxx.xxx.xxx.xxx
> a=sendrecv
> a=rtpmap:31 H261/90000
> a=fmtp:31 cif=1;qcif=1
>
> --- (13 headers 21 lines)---
> Sending to xxx.xxx.xxx.xxx : 5060 (NAT)
> Using INVITE request as basis request -  
> 9c00870094007300 at xxx.xxx.xxx.xxx
> Found user '3005'
> Found RTP audio format 96
> Found RTP audio format 97
> Found RTP audio format 9
> Found RTP audio format 8
> Found RTP audio format 0
> Found RTP video format 31
> Peer audio RTP is at port xxx.xxx.xxx.xxx:5600
> Found description format G7221 for ID 96
> Found description format G7221 for ID 97
> Found description format G722 for ID 9
> Found description format PCMA for ID 8
> Found description format PCMU for ID 0
> Found description format H261 for ID 31
> Capabilities: us - 0x28070f (g723|gsm|ulaw|alaw|g729|speex|ilbc| 
> h263|h264), peer - audio=0x4040c (ulaw|alaw|ilbc|h261)/ 
> video=0x40000 (h261), combined - 0x40c (ulaw|alaw|ilbc)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer -  
> 0x0 (nothing), combined - 0x0 (nothing)
> Peer audio RTP is at port xxx.xxx.xxx.xxx:5600
> Peer video RTP is at port xxx.xxx.xxx.xxx:5602
> Looking for 3001 in from-sip (domain aaa.aaa.aaa.aaa)
> list_route: hop: <sip:3005 at xxx.xxx.xxx.xxx>
> Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bK184b70b8ed4c2761877486319683c457.1;rport;received=x 
> xx.xxx.xxx.xxx
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:3001 at aaa.aaa.aaa.aaa>
> Content-Length: 0
>
>
> ---
>     -- Executing [3001 at from-sip:1] Dial("SIP/3005-08dd1948", "SIP/ 
> 3001|20") in new stack
>     -- Executing [3001 at from-sip:1] Dial("SIP/3005-08dd1948", "SIP/ 
> 3001|20") in new stack
> Video is at aaa.aaa.aaa.aaa port 11862
> Audio is at aaa.aaa.aaa.aaa port 17218
> Video is at aaa.aaa.aaa.aaa port 11862
> Audio is at aaa.aaa.aaa.aaa port 17218
> Adding codec 0x8 (alaw) to SDP
> Adding codec 0x400 (ilbc) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to yyy.yyy.yyy.yyy:5060:
> INVITE sip:yyy.yyy.yyy.yyy SIP/2.0
> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK0fcd45d2;rport
> From: "BK" <sip:3005 at aaa.aaa.aaa.aaa>;tag=as0294a8fa
> To: <sip:yyy.yyy.yyy.yyy>
> Contact: <sip:3005 at aaa.aaa.aaa.aaa>
> Call-ID: 4810f285311c63fc642bc5b86fd3f168 at aaa.aaa.aaa.aaa
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Wed, 05 Jul 2006 21:12:42 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 274
>
> v=0
> o=root 11515 11515 IN IP4 xxx.xxx.xxx.xxx
> s=session
> c=IN IP4 xxx.xxx.xxx.xxx
> t=0 0
> m=audio 5600 RTP/AVP 8 97 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=sendrecv
>
> ---
> Adding codec 0x8 (alaw) to SDP
> Adding codec 0x400 (ilbc) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
>     -- Called 3001
> Reliably Transmitting (no NAT) to yyy.yyy.yyy.yyy:5060:
> INVITE sip:yyy.yyy.yyy.yyy SIP/2.0
> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK0fcd45d2;rport
> From: "BK" <sip:3005 at aaa.aaa.aaa.aaa>;tag=as0294a8fa
> To: <sip:yyy.yyy.yyy.yyy>
> Contact: <sip:3005 at aaa.aaa.aaa.aaa>
> Call-ID: 4810f285311c63fc642bc5b86fd3f168 at aaa.aaa.aaa.aaa
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Wed, 05 Jul 2006 21:12:42 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 274
>
> v=0
> o=root 11515 11515 IN IP4 xxx.xxx.xxx.xxx
> s=session
> c=IN IP4 xxx.xxx.xxx.xxx
> t=0 0
> m=audio 5600 RTP/AVP 8 97 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=sendrecv
>
> ---
>     -- Called 3001
> nyguest25*CLI>
> <-- SIP read from yyy.yyy.yyy.yyy:5060:
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK0fcd45d2;rport
> Call-ID: 4810f285311c63fc642bc5b86fd3f168 at aaa.aaa.aaa.aaa
> CSeq: 102 INVITE
> From: "BK" <sip:3005 at aaa.aaa.aaa.aaa>;tag=as0294a8fa
> To: <sip:yyy.yyy.yyy.yyy>;tag=3f5d9699382bf3f3
> Allow: UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
> Server: TANDBERG/49 (F4.1 NTSC)
> Content-Length:0
>
>
> --- (9 headers 0 lines)---
>
> <-- SIP read from yyy.yyy.yyy.yyy:5060:
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK0fcd45d2;rport
> Call-ID: 4810f285311c63fc642bc5b86fd3f168 at aaa.aaa.aaa.aaa
> CSeq: 102 INVITE
> From: "BK" <sip:3005 at aaa.aaa.aaa.aaa>;tag=as0294a8fa
> To: <sip:yyy.yyy.yyy.yyy>;tag=3f5d9699382bf3f3
> Allow: UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
> Server: TANDBERG/49 (F4.1 NTSC)
> Content-Length:0
>
>
> --- (9 headers 0 lines)---
>     -- SIP/3001-08dd9bf8 is ringing
> Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bK184b70b8ed4c2761877486319683c457.1;rport;received=x 
> xx.xxx.xxx.xxx
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>;tag=as2d4b6119
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:3001 at aaa.aaa.aaa.aaa>
> Content-Length: 0
>
>
> ---
>     -- SIP/3001-08dd9bf8 is ringing
> Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bK184b70b8ed4c2761877486319683c457.1;rport;received=x 
> xx.xxx.xxx.xxx
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>;tag=as2d4b6119
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:3001 at aaa.aaa.aaa.aaa>
> Content-Length: 0
>
>
> ---
> nyguest25*CLI> Jul  5 17:12:44 WARNING[11545]: chan_sip.c:1828  
> retrans_pkt: Maximum retries exceeded on transmission  
> 0f8522eb2ae0624b13e179fb582023de at aaa.aaa.aaa.aaa for seqno 102 (Non- 
> critical Request)
> Jul  5 17:12:44 WARNING[11545]: chan_sip.c:1828 retrans_pkt:  
> Maximum retries exceeded on transmission  
> 0f8522eb2ae0624b13e179fb582023de at aaa.aaa.aaa.aaa for seqno 102 (Non- 
> critical Request)
> nyguest25*CLI>
> <-- SIP read from yyy.yyy.yyy.yyy:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK0fcd45d2;rport
> Call-ID: 4810f285311c63fc642bc5b86fd3f168 at aaa.aaa.aaa.aaa
> CSeq: 102 INVITE
> Contact: <sip:yyy.yyy.yyy.yyy>
> From: "BK" <sip:3005 at aaa.aaa.aaa.aaa>;tag=as0294a8fa
> To: <sip:yyy.yyy.yyy.yyy>;tag=3f5d9699382bf3f3
> Allow: UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
> Server: TANDBERG/49 (F4.1 NTSC)
> Content-Type:application/sdp
> Content-Length:201
>
> v=0
> o=tandberg 0 1 IN IP4 yyy.yyy.yyy.yyy
> s=-
> c=IN IP4 yyy.yyy.yyy.yyy
> b=CT:768
> t=0 0
> m=audio 5600 RTP/AVP 8
> c=IN IP4 yyy.yyy.yyy.yyy
> b=TIAS:64000
> a=sendrecv
> a=rtpmap:8 PCMA/8000
> a=maxprate:50.0
>
> --- (11 headers 12 lines)---
> Found RTP audio format 8
> Peer audio RTP is at port yyy.yyy.yyy.yyy:5600
> Found description format PCMA for ID 8
> Capabilities: us - 0x380408 (alaw|ilbc|h263|h263p|h264), peer -  
> audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer -  
> 0x0 (nothing), combined - 0x0 (nothing)
> Peer audio RTP is at port yyy.yyy.yyy.yyy:5600
> list_route: hop: <sip:yyy.yyy.yyy.yyy>
> set_destination: Parsing <sip:yyy.yyy.yyy.yyy> for address/port to  
> send to
>
> <-- SIP read from yyy.yyy.yyy.yyy:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK0fcd45d2;rport
> Call-ID: 4810f285311c63fc642bc5b86fd3f168 at aaa.aaa.aaa.aaa
> CSeq: 102 INVITE
> Contact: <sip:yyy.yyy.yyy.yyy>
> From: "BK" <sip:3005 at aaa.aaa.aaa.aaa>;tag=as0294a8fa
> To: <sip:yyy.yyy.yyy.yyy>;tag=3f5d9699382bf3f3
> Allow: UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
> Server: TANDBERG/49 (F4.1 NTSC)
> Content-Type:application/sdp
> Content-Length:201
>
> v=0
> o=tandberg 0 1 IN IP4 yyy.yyy.yyy.yyy
> s=-
> set_destination: set destination to yyy.yyy.yyy.yyy, port 5060
>  IP4 yyy.yyy.yyy.yyy
> b=CT:768
> t=0 0
> m=audio 5600 RTP/AVP 8
> c=IN IP4 yyy.yyy.yyy.yyy
> b=TIAS:64000
> a=sendrecv
> a=rtpmap:8 PCMA/8000
> a=maxprate:50.0
>
> --- (11 headers 12 lines)---
> Found RTP audio format 8
> Peer audio RTP is at port yyy.yyy.yyy.yyy:5600
> Found description format PCMA for ID 8
> Capabilities: us - 0x380408 (alaw|ilbc|h263|h263p|h264), peer -  
> audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
> Transmitting (no NAT) to yyy.yyy.yyy.yyy:5060:ephone-event), peer -  
> 0x0 (nothing), combined -
> ACK sip:yyy.yyy.yyy.yyy SIP/2.0
> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK60a3345b;rport
> From: "BK" <sip:3005 at aaa.aaa.aaa.aaa>;tag=as0294a8fa
> To: <sip:yyy.yyy.yyy.yyy>;tag=3f5d9699382bf3f3
> Contact: <sip:3005 at aaa.aaa.aaa.aaa>
> Call-ID: 4810f285311c63fc642bc5b86fd3f168 at aaa.aaa.aaa.aaa
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
> ---
>     -- SIP/3001-08dd9bf8 answered SIP/3005-08dd1948
>  0x0 (nothing)
> Peer audio RTP is at port yyy.yyy.yyy.yyy:5600
> list_route: hop: <sip:yyy.yyy.yyy.yyy>
> set_destination: Parsing <sip:yyy.yyy.yyy.yyy> for address/port to  
> send to
> set_destination: set destination to yyy.yyy.yyy.yyy, port 5060
> Transmitting (no NAT) to yyy.yyy.yyy.yyy:5060:
> ACK sip:yyy.yyy.yyy.yyy SIP/2.0
> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK60a3345b;rport
> From: "BK" <sip:3005 at aaa.aaa.aaa.aaa>;tag=as0294a8fa
> To: <sip:yyy.yyy.yyy.yyy>;tag=3f5d9699382bf3f3
> Contact: <sip:3005 at aaa.aaa.aaa.aaa>
> Call-ID: 4810f285311c63fc642bc5b86fd3f168 at aaa.aaa.aaa.aaa
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
> ---
>     -- SIP/3001-08dd9bf8 answered SIP/3005-08dd1948
> Audio is at aaa.aaa.aaa.aaa port 15984
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x8 (alaw) to SDP
> Adding codec 0x400 (ilbc) to SDP
> Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bK184b70b8ed4c2761877486319683c457.1;rport;received=x 
> xx.xxx.xxx.xxx
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>;tag=as2d4b6119
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:3001 at aaa.aaa.aaa.aaa>
> Content-Type: application/sdp
> Content-Length: 242
>
> v=0
> o=root 11515 11515 IN IP4 yyy.yyy.yyy.yyy
> s=session
> c=IN IP4 yyy.yyy.yyy.yyy
> t=0 0
> m=audio 5600 RTP/AVP 0 8 97
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=20
> a=silenceSupp:off - - - -
> a=sendrecv
>
> ---
>     -- Native bridging SIP/3005-08dd1948 and SIP/3001-08dd9bf8
> Audio is at aaa.aaa.aaa.aaa port 15984
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x8 (alaw) to SDP
> Adding codec 0x400 (ilbc) to SDP
> Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bK184b70b8ed4c2761877486319683c457.1;rport;received=x 
> xx.xxx.xxx.xxx
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>;tag=as2d4b6119
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:3001 at aaa.aaa.aaa.aaa>
> Content-Type: application/sdp
> Content-Length: 242
>
> v=0
> o=root 11515 11515 IN IP4 yyy.yyy.yyy.yyy
> s=session
> c=IN IP4 yyy.yyy.yyy.yyy
> t=0 0
> m=audio 5600 RTP/AVP 0 8 97
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=20
> a=silenceSupp:off - - - -
> a=sendrecv
>
> ---
>     -- Native bridging SIP/3005-08dd1948 and SIP/3001-08dd9bf8
> nyguest25*CLI>
> <-- SIP read from xxx.xxx.xxx.xxx:5060:
> ACK sip:3001 at aaa.aaa.aaa.aaa SIP/2.0
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bK184b70b8ed4c2761877486319683c457.1;rport
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 101 ACK
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>;tag=as2d4b6119
> Max-Forwards: 70
> User-Agent: TANDBERG/46 (F3.3 NTSC)
> Content-Length:0
>
>
> --- (9 headers 0 lines)---
>
> <-- SIP read from xxx.xxx.xxx.xxx:5060:
> ACK sip:3001 at aaa.aaa.aaa.aaa SIP/2.0
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bK184b70b8ed4c2761877486319683c457.1;rport
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 101 ACK
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>;tag=as2d4b6119
> Max-Forwards: 70
> User-Agent: TANDBERG/46 (F3.3 NTSC)
> Content-Length:0
>
>
> --- (9 headers 0 lines)---
> Really destroying SIP dialog  
> '0f8522eb2ae0624b13e179fb582023de at aaa.aaa.aaa.aaa' Method: NOTIFY
> Really destroying SIP dialog  
> '0f8522eb2ae0624b13e179fb582023de at aaa.aaa.aaa.aaa' Method: NOTIFY
> Really destroying SIP dialog  
> '1bc2376b544379b8485b619578463123 at aaa.aaa.aaa.aaa' Method: NOTIFY
> Really destroying SIP dialog  
> '1bc2376b544379b8485b619578463123 at aaa.aaa.aaa.aaa' Method: NOTIFY
> nyguest25*CLI>
> <-- SIP read from xxx.xxx.xxx.xxx:5060:
> BYE sip:3001 at aaa.aaa.aaa.aaa SIP/2.0
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bK1f5656448fb8f79f4f6dd7f81e4f2fe2.1;rport
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 102 BYE
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>;tag=as2d4b6119
> Max-Forwards: 70
> User-Agent: TANDBERG/46 (F3.3 NTSC)
> Content-Length:0
>
>
> --- (9 headers 0 lines)---
> Sending to xxx.xxx.xxx.xxx : 5060 (NAT)
> Transmitting (NAT) to xxx.xxx.xxx.xxx:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bK1f5656448fb8f79f4f6dd7f81e4f2fe2.1;received=xxx.xxx 
> .xxx.xxx;rport=5060
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>;tag=as2d4b6119
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 102 BYE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:3001 at aaa.aaa.aaa.aaa>
> Content-Length: 0
>
>
> ---
>
> <-- SIP read from xxx.xxx.xxx.xxx:5060:
> BYE sip:3001 at aaa.aaa.aaa.aaa SIP/2.0
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bK1f5656448fb8f79f4f6dd7f81e4f2fe2.1;rport
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 102 BYE
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>;tag=as2d4b6119
> Max-Forwards: 70
> User-Agent: TANDBERG/46 (F3.3 NTSC)
> Content-Length:0
>
>
> --- (9 headers 0 lines)---
> Sending to xxx.xxx.xxx.xxx : 5060 (NAT)
> Transmitting (NAT) to xxx.xxx.xxx.xxx:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 
> 5060;branch=z9hG4bK1f5656448fb8f79f4f6dd7f81e4f2fe2.1;received=xxx.xxx 
> .xxx.xxx;rport=5060
> From: "BK" <sip: 
> 3005 at aaa.aaa.aaa.aaa>;tag=09002500ea00d400;epid=TAA00506001DE48
> To: <sip:3001 at aaa.aaa.aaa.aaa>;tag=as2d4b6119
> Call-ID: 9c00870094007300 at xxx.xxx.xxx.xxx
> CSeq: 102 BYE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:3001 at aaa.aaa.aaa.aaa>
> Content-Length: 0
>
> set_destination: Parsing <sip:yyy.yyy.yyy.yyy> for address/port to  
> send to
> nyguest25*CLI>
> ---
> set_destination: Parsing <sip:yyy.yyy.yyy.yyy> for address/port to  
> send to
> set_destination: set destination to yyy.yyy.yyy.yyy, port 5060
> Reliably Transmitting (no NAT) to yyy.yyy.yyy.yyy:5060:
> BYE sip:yyy.yyy.yyy.yyy SIP/2.0
> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK64b8feae;rport
> From: "BK" <sip:3005 at aaa.aaa.aaa.aaa>;tag=as0294a8fa
> To: <sip:yyy.yyy.yyy.yyy>;tag=3f5d9699382bf3f3
> Contact: <sip:3005 at aaa.aaa.aaa.aaa>
> Call-ID: 4810f285311c63fc642bc5b86fd3f168 at aaa.aaa.aaa.aaa
> CSeq: 103 BYE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
> ---
>   == Spawn extension (from-sip, 3001, 1) exited non-zero on 'SIP/ 
> 3005-08dd1948'
> set_destination: set destination to yyy.yyy.yyy.yyy, port 5060
> Reliably Transmitting (no NAT) to yyy.yyy.yyy.yyy:5060:
> BYE sip:yyy.yyy.yyy.yyy SIP/2.0
> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK64b8feae;rport
> From: "BK" <sip:3005 at aaa.aaa.aaa.aaa>;tag=as0294a8fa
> To: <sip:yyy.yyy.yyy.yyy>;tag=3f5d9699382bf3f3
> Contact: <sip:3005 at aaa.aaa.aaa.aaa>
> Call-ID: 4810f285311c63fc642bc5b86fd3f168 at aaa.aaa.aaa.aaa
> CSeq: 103 BYE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
> ---
>   == Spawn extension (from-sip, 3001, 1) exited non-zero on 'SIP/ 
> 3005-08dd1948'
> nyguest25*CLI>
> <-- SIP read from yyy.yyy.yyy.yyy:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK64b8feae;rport
> Call-ID: 4810f285311c63fc642bc5b86fd3f168 at aaa.aaa.aaa.aaa
> CSeq: 103 BYE
> From: "BK" <sip:3005 at aaa.aaa.aaa.aaa>;tag=as0294a8fa
> To: <sip:yyy.yyy.yyy.yyy>;tag=3f5d9699382bf3f3
> Allow: UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
> Server: TANDBERG/49 (F4.1 NTSC)
> Content-Length:0
>
>
> --- (9 headers 0 lines)---
> Really destroying SIP dialog  
> '4810f285311c63fc642bc5b86fd3f168 at aaa.aaa.aaa.aaa' Method: INVITE
> Really destroying SIP dialog '9c00870094007300 at xxx.xxx.xxx.xxx'  
> Method: BYE
>
> <-- SIP read from yyy.yyy.yyy.yyy:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK64b8feae;rport
> Call-ID: 4810f285311c63fc642bc5b86fd3f168 at aaa.aaa.aaa.aaa
> CSeq: 103 BYE
> From: "BK" <sip:3005 at aaa.aaa.aaa.aaa>;tag=as0294a8fa
> To: <sip:yyy.yyy.yyy.yyy>;tag=3f5d9699382bf3f3
> Allow: UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
> Server: TANDBERG/49 (F4.1 NTSC)
> Content-Length:0
>
>
> --- (9 headers 0 lines)---
> Really destroying SIP dialog  
> '4810f285311c63fc642bc5b86fd3f168 at aaa.aaa.aaa.aaa' Method: INVITE
> Really destroying SIP dialog '9c00870094007300 at xxx.xxx.xxx.xxx'  
> Method: BYE
>
>
>
> On Jul 5, 2006, at 5:13 PM, John Martin wrote:
>
>> "show codecs" has a "bug" in it (even in trunk) where it doesn't  
>> go far
>> enough to display the H.264 codec.
>>
>> What version of Asterisk are you using?
>>
>> What's the endpoint you're using - some don't like the way that  
>> Asterisk
>> currently don't send fmtp in the SDP.
>>
>> John Martin
>> http://www.AuPix.com
>>

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